similar to: Zultys 4x4 or 4x5 ip phones?

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

hi, im looking at deploying asterisk in a small corporate enviroment which will have approx. 1200 IP Phones and an average of about 100 to 200 calls at any given time. The calls will be sent out SIP to my Cisco Gateway. Im running Asterisk on a Dell Dual P3 1.2ghz running Fedora. Is there a calculator or a spreadsheet which could tell me about how many calls I will be able to make through *

Hi All, A friend has asked me to get a new Zultys 4x5 working with his * server. I've been over the web gui but don't see where all the registration info gets entered in the phone. Is there someone on-list who has Zultys phones and can advise me as to their setup? The manufacturers docs are useless. Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr.

Hi peoples Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they have configured one before for iax. If you have a sample config file that would be great. Any assistance would be nice Scott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/6c243053/attachment.htm

Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we

There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

Good Day List, I have my asterisk box setup to be an ntp server, and my zultys 4X4 phone is able to get the time, however I must first select the TimeZone Offset and then it will use the time setting from my server. This is a hassle because every time the phone reboots the user must answer this question and as you can imagine End users do not know what to do and since the phone is not

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

Anyone had experience with this phone? The interesting features that caught my eye were the RTP encryption, speaker "pager", and built in 4-port ethernet switch. Of course, RTP encryption doesn't exist in * yet, but it might be interesting (I've had people ask about it, but I don't know how serious they are about needing it.) http://www.zip4x4.com/summary_ZIP4x4.htm

Does anyone on-list have any advice about the version 2.2 firmware for the Zultys 4x5 phone? It has a new "gateway" mode that is supposed to direct calls on the analogue line to the PBX for VM, etc. Zultys has not yet presented any docs, and the keep refering me to the reseller who doesn't know anything about t at all. Thanks, Michael -- Michael Graves

Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best "enterprise" options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about

Hi All, Does anyone on-list have experience with the Zultys 4x4 and 4x5 phones with AES encryption? Does this work well with *? I imagine that it would work as long as * doesn't have to decode the stream. What are the implications for calls going to voicemail or MeetMe? I'd pondering installing the 4x5, especially if Zultys can make the on-board FXO port available to * as well as a

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

Hi all, I have found the solution for getting Zultys handsets to accept paging messages. We need an app to send RTP packets with a codec of G711U to send to port 3771 on multicast dest of 224.0.0.1. There isn't any need to use SDP or SCCP packets to inform the phone to "listen" for those packets, they do it by default. Finally we may be able to get a solution going for Zultys

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

I'm working on getting two SIP servers to talk. An Asterisk box and a Zultys MX250 system. So far, things have been working pretty well. I can call from one of my Asterisk-managed phones to an MX250-managed phone and vice versa. However, there are some strange issues. If I call from an MX250 phone to an Asterisk phone, the conversation is ok, but there is a noticeable delay in the voice

Has anyone who's gotten this message managed to figure it out and fix it? I've been scouring the mailing list for clues but I'm still no closer. I have 2 asterisk servers, old and new. I'm trying to switch to the new server. I am using a 2.4 kernel on the new and a 2.6 on the old. I am running 0.9.0 on the old and my sip phones work fine, on the new Ihave tried 0.9.0,

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

This just started happening today. I've got 1 queue and 6 agents. All logged in. I tell the service people to ignore my call if they see my caller id. I call the queue and watch as asterisk bounces me around the phones. Our agent ring time is 5 second timeout and a 5 second wait time before trying next agent. I get the same message in console for each agent attempt: -- Executing

This is probably going to sound really silly and I must be confused about it. Maybe someone can set me straight. I've been tinkering for a while with * and a number of different FXO/FXS cards, SIP phones, and ATAs trying to get a feel for what works and what doesn't. In the SIP phone group, I have a Cisco 7940G, a Polycom 500, and now an SPA-841. Each of these allow me to configure at